The present invention concerns an audio-intonation calibration method.
It also concerns a method of practicing speaking a language being studied by a subject and a method of performing a song by a subject.
Generally speaking, the present invention concerns a method in which the emission of an audio signal by a subject is modified by modifying the sound information that he receives when he speaks.
A method of this kind is based on a principle known in the art whereby the vocal provision of a subject, i.e. sounds that he emits, undergoes a major transformation as a function of the auditory provision applied to the same subject, i.e. sound information that he receives.
Using equipment in which an audio signal emitted by a subject is reproduced to the auditory organs of the subject after real time processing is known in the art, and especially in the particular field of teaching and practicing speaking languages.
A method of this kind is described in the document WO 92/14229 in particular.
That document describes a device in which an audio signal emitted by a subject is modified by processing it to take account of the characteristics of a foreign language being studied and of the harmonic content of that language. The modified audio signal is then furnished to the subject in real time by a vibratory signal, generally by a sound signal, in order to modify the audio signal emitted by the subject.
However, in the above document, the audio signal emitted by the subject is processed in a predetermined manner as a function of the bandwidth of the language being learned, and in particular as a function of the envelope curve of that bandwidth.
In practice, the signal processing circuit comprises a multifrequency equalizer that is adjusted in a predetermined manner as a function of the foreign language in question, and more particularly of the bandwidth of that language and the shape of that bandwidth, i.e. the envelope curve of the bandwidth. In practice, the equalizer consists of a plurality of successive filters set to different frequencies.
Thus frequency parameters of the equalizer are set in a predetermined manner by characteristics of the language being studied.
Similarly, the above document describes a second form of processing applied to the audio signal emitted by the subject in which adjustment parameters are established as a function of the sound vibration harmonic content of the utterance collected and parameters depending on the language being studied.
In this case, the parameters of a multifrequency equalizer are set by the difference between the processed signal coming from the effective vocal provision of the subject and predetermined characteristics of the language.
Thus the above document describes the creation from a first signal representative of the sound emitted by the subject of a second signal modified with respect to the first signal in a predetermined manner and as a function of the bandwidth of the language being learned, and in particular as a function of the envelope curve of that language, and a third signal which is derived from the first signal by modifying it as a function of the effective harmonic content of the utterance and characteristics of the language.
The signals are then selectively reproduced to the subject.
The above kind of prior art system has the drawback of using predefined parameter settings, taking no account of the type of audio signal emitted by the subject.